# Configuring WebRTC in Asterisk (FreePBX) ## Technical requirements * **The telephony server must be accessible from the Internet, i.e. it must have a static IP.** * **Must have a working SSL/TLS certificate.** * Asterisk version not less than 11 ([?](#webrtc-support-in-asterisk)), but for convenience preferably 15 ([?](#pjsip-in-asterisk)) or at least 13 ([?](#configuration-examples)). * When using browser-based softphone, wss (WebSocket Secure) must be configured on the Asterisk server, and the port must be open to the outside (usually 8089) ([?](#freepbx)). This is configured in the admin interface: Advanced Settings > Asterisk Builtin mini-HTTP section > HTTPS Bind Port * Port range for RTP (typically 10000-20000) ([?](#freepbx)). It is configured in the administrator interface: Settings → Asterisk SIP Settings → General SIP Settings Tab ## Setting This example describes how to configure WebRTC in an already running FreePBX server: * is available at \.com and **has an SSL/TLS certificate**; * FreePBX version 16, Asterisk version 15; * wss is configured on port 8089; * all other relevant ports are open. ## Configuring FreePBX {% hint style="danger" %} If the "Change To CHAN\_PjSIP Driver" button (see below, in the internal number setting) is available, you do not need to do anything in this section. {% endhint %} **Settings ➝ Advanced Settings** | Section Dialplan and Operational | | | -------------------------------- | ---- | | SIP Channel Driver | both | **Settings ➝ Asterisk SIP Settings** **Tab “SIP Legacy Settings \[chan\_sip]”** **Verify that the variables are set:** | Section Advanced General Settings | | | --------------------------------- | ----------------------------------------------------------------------------------- | | Other SIP Settings |

icesupport = yes

media\_encryption = dtls

tlscipher = AES256-SHA

| ## Creating and setting up an extension number (extension) Go to the FreePBX configuration interface and log in: https\://\.com/ Go to section **Applications ➝ Extensions** Button “**+ Add Extension**” **➝** “**+ Add New SIP (Legacy) \[chan\_sip] Extension**” (other versions of FreePBX may be “Add New Chan\_SIP extension”) **In the tab General:** | Section Add Extension | | | --------------------- | ---- | | User Extension | 1001 | | Display Name | 1001 | **Press the button Submit** After that, open editing of the newly created extension in the list of extensions (in the line with 5001 in the Actions column, click the edit button with a pencil icon). **Go to the tab Advanced** **First change the SIP Driver to PjSIP:** | Tab Advanced → Section Edit Extension | | | ------------------------------------- | --------------------------- | | Enable WebRTC defaults | Yes | | Media Encryption | DTLS-SRTP (not recommended) | **Press the button Submit** **Then change the SIP Driver back:** | Tab Advanced → Section Edit Extension | | | ------------------------------------- | -------------------------- | | Change SIP Driver | Change To CHAN\_SIP Driver | **Now to make the remaining customizations:** | Tab Advanced → Section Edit Extension | | | ------------------------------------- | --------------------- | | Transport | All - WSS Primary | | Enable AVPF | Yes | | Force AVP | Yes | | Enable ICE Support | Yes | | Enable rtcp Mux | Yes | | Enable Encryption | Yes (SRTP only) | | Section DTLS | | | Enable DTLS | Yes (there should be) | | Use Certificate | \.com | **Press the button Submit** **Press the button Apply Config** ## Testing that wss works Let's use the sipML5 live demo from Doubango as an example: ### Setting **First, the "Expert mode?" button.** The advanced settings page should open in the next tab | Disable Video | v (checkmark) | | -------------------- | ------------------- | | WebSocket Server URL | wss\://.com:8089/ws | **Press the button Save** **Back to the main tab** | Display Name | 1001 | | ---------------- | ----------------------------------------------------------------------------------------------- | | Private Identity | 1001 | | Public Identity | sip:1001@\.com | | Password |

Copy from FreePBX settings

Extension: 1001 ➝ General tab ➝ Edit Extension ➝ Secret

| | Realm | \.com | **Button Login** The word Connected should appear at the top above the Registration heading. ### Call Now you can make the call. Under the Call control heading, dial the desired number (\*43 for an echo call or a cell phone number with 8 first). **Press the button Call ➝ Audio** **There's a call that should go out.** **Profit!** ### Links #### **WebRTC support in Asterisk** Initial support for WebRTC in Asterisk starting with version 11: * [New in 11 - Asterisk Project - Asterisk Project Wiki](https://wiki.asterisk.org/wiki/display/AST/New+in+11) * [Asterisk WebRTC Support - Asterisk Project - Asterisk Project Wiki](https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support) #### **PjSIP in Asterisk** * New chan\_pjsip driver in Asterisk 12: [New in 12 - Asterisk Project - Asterisk Project Wiki](https://wiki.asterisk.org/wiki/display/AST/New+in+12) * PjSIP is available by default in Asterisk 15: [PJSIP-pjproject - Asterisk Project - Asterisk Project Wiki](https://wiki.asterisk.org/wiki/display/AST/PJSIP-pjproject) #### **FreePBX** * [Ports used on your PBX - PBX Platforms - Documentation](https://wiki.freepbx.org/display/PPS/Ports+used+on+your+PBX) #### **Configuration examples** * [Configuring Asterisk for WebRTC Clients](https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients) * [WebRTC clients SIPML5 JSSIP setup configurations and complete end to end client server call demo](https://youtu.be/7pA7j0kt5vA)